similar to: No DTMF interpretation on outgoing calls

Does the SugarCRM included with AAH 1.3 not meet this criteria for you? ------------------------------ Message: 3 Date: Tue, 26 Jul 2005 17:48:08 +0100 From: "JunkMail" <junkmail@segurajuda.dyndns.org> Subject: [Asterisk-Users] Dial using URI(web) or using FORM(web) To: <asterisk-users@lists.digium.com> Message-ID: <009101c59201$cd5cc9c0$0a00a8c0@segurajuda.local>

Hi all! After much struggling I got my *@home working fine AND making use of two AVMFritz!PCI cards. Really nice ! (kernel 2.4.2x) There's however a silly glitch that's getting on my nerves, and, kind of a newbie that I am to linux, it should be easy to get help : -- "capiinit start" MUST BE run before Asterisk. (any other way makes * not to start because chan_capi

If I want to send outgoing DTMF over a BRI interface, can I do it with 'isdn4linux' or must I use the 'capi' library? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ mje@posix.co.za - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 -------------- next part -------------- A non-text attachment was

Ok, this falls under the newbie category: Has anybody created any documentation on using musiconhold or call parking? I have found sample config files for musiconhold, but I'm not sure how they work. [musiconhold.conf] [classes] loud=>mp3:/var/lib/asteriks/mohmp3 How do I then set up this feature in extensions.conf? I can't seem to find an example of what I'm looking for (or I

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

Hello I almost managed to make everything work my server isn't in prodution because i had some problem on others system but last week i come back and i run an update on ubuntu 12.04.1 to 12.04.2 I don't known if its' the reaseon or if someone has change something but scripts no longer work I have a message "returned 2" Can you help me because I don't find how to solve

Folks, I'm setting up to sync files from a staging env to a prodution env, such that the filesystems for both are served via nfs to their respective systems. Thus, I can do the rsync on the nfs server as a 'advanced cp'. rsnyc $parms filessystem filesystem2 Due to ownerships and permissions, I have to run this as root, but, I'd like the files to belong to another user and

Not sure what is going on here, we are trying to push out a new version of one of our tools to production. Everything works perfectly in development (mac) and on our staging server (debian sarge) but the prodution server (debian sarge) hates what we have done. We have some variables set in the various environments *.rb file One of them is a link to another product. It changes based on what

Hi, I have a program that writes lots of files to a directory tree (around 15 Million fo files), and a node can have up to 400000 files (and I don't have any way to split this ammount in smaller ones). As the number of files grows, my application gets slower and slower (the app is works something like a cache for another app and I can't redesign the way it distributes files into disk due

Hello all, This afternoon I had a BRI line installed by Telstra (our telco in Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux driver. Incoming and outgoing calls with Asterisk work fine (and with no echo - my main reason for getting ISDN). However, I can't seem to get outgoing DTMF working (incoming works fine). I made a call from my desk phone (Cisco 7940G)

The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. However, this presents another problem. When I'm using g729 to place a call, I get the warning "Unable to process inband DTMF" because inband is not supposed to work with g729 (although it does seem to work when I've tried it so far).

Hi, I've got a problem with some grandstram devices (namely a couple of budgetone 101 and an ata-486). The point is that, unless I use inband for DTMF, asterisk ignore the first digit dialed. Inband DTMF forces me to use A-law/Mu-law, which is not what I want. BTW, this appens after a Playtones(), waiting for user entering an extension. I've tried many solutions, played around with

I am using a grandstream phone with g729 and alaw odecs and in both modes I cannot seem to pass dtmf's, neither inband nor out of band, neither wthrough a lcoal server nor through a natted connection. Am I missing something ?

Hi, I would like to use Asterisk as VoiceMail system ... the only issue I have is with DTMF recognition. Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ... inband : works, but has some glitch ... not always good ... don't know why. rfc2833 : doesn't seem to work .. info : said to be not working

Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working. Any help on why it breaks? Any possible fix? /tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c 289d288 < int capability; 3921,3922d3919 < p->capability = user->capability;

I've got a problem with DTMF, again. My asterisk box is connected with the outside world (PSTN) via a sip proxy. The problem is that for some reason, I need to use rfc2833 for signaling digits to the gateway and inband to accept digits from outside (eg. when someone dials one of our DIDs). It's possible to do this? I've ever tried splitting 'peer' and 'user' part in

Hello, I'm trying to understand why when I make a call from a SIP phone to an external number who has an IVR system in which I've to choose some options using the dialpad, it does'nt recognise the key pressed and remains still waiting for my choose. I'm tryng using Grandstream 102, and i've tryed with all the 3 modes possibile: Dtmf inband, rfc2833 and INFO (obviously

Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

Sorry if this question already asked. I'm implementing Voip with asterisk and grandstream gxw4108, according from the manual, for connecting with PSTN I must configure one SIP account and assign that for dialing the PSTN so in my sip.conf I configure SIP account(extension) : [1401] type=friend username=1401 secret=1401 host=dynamic context=my-office insecure=port in my extension.con